Issues 09000 - 09999

[..]
ASTERISK-09000: correct response sent in local_attended_transfer()?
ASTERISK-09001: VoiceMail() broadcasts limited to 256 bytes
ASTERISK-09002: console disconnects on reload
ASTERISK-09003: [patch] dumphtml for ami commands
ASTERISK-09004: [patch] hide secret from manager show user foo
ASTERISK-09005: [patch] Improper gain param syntax in app_voicemail causes segfault
ASTERISK-09006: heavy traffic in postgresql cdr database causes PRI errors
ASTERISK-09007: safe asterisk script crashes with bad file descriptor, asterisk doesnt start at boot
ASTERISK-09008: [patch] sip set debug on got lost somewhere
ASTERISK-09009: Random Crashes on IAX2 trunk
ASTERISK-09010: [patch] fields "start", "answer" and "end" are always empty.
ASTERISK-09011: SpeechBackground features
ASTERISK-09012: [patch] Adding fxotune to man pages
ASTERISK-09013: [patch] Half Duplex Audio on SIP Calls Using MixMonitor/AMD
ASTERISK-09014: [patch] allow call to continue after agent hangs up
ASTERISK-09015: AGI does not report when script cannot be found/executed
ASTERISK-09016: Crash on dereferencing null pointer in chan_zap.c - zt_get_index()
ASTERISK-09017: CDR variables are not used correctly after a FORKCDR
ASTERISK-09018: DTMF detection on IAX clients / Asterisk 1.4.1
ASTERISK-09019: Problem with connecting asterisk with a TE210 board MCI truncs running NFAS
ASTERISK-09020: Extension length limit applied to Dial arguments
ASTERISK-09021: Zaptel crash in hpec_channel_update
ASTERISK-09022: Translator paths are unregistered on a reload for the TC400P
ASTERISK-09023: Asterisk crashes random with iax2
ASTERISK-09024: Typo in func_cdr.c causes help text to display improperly
ASTERISK-09025: README.Linux26 still distributed with 1.2.16
ASTERISK-09026: Channel hung up immediately after successful SMS
ASTERISK-09027: Zap device not set up
ASTERISK-09028: segfault after a while of operation (maybe a month)
ASTERISK-09029: Bad zapata.conf configuration crashes asterisk
ASTERISK-09030: app_voicemail changes in r58931
ASTERISK-09031: ChanSpy() stops working when spied channel starts Record()
ASTERISK-09032: [patch] disable building codec_zap for zaptel older than 1.2.13
ASTERISK-09033: [patch] Update help text to match functionality for sip debug
ASTERISK-09034: [patch] REINVITE before 200ok causes a call to be ended
ASTERISK-09035: waitforsilence still timesout inappropriately
ASTERISK-09036: Asterisk 1.4.1 FollowMe not continuing to next priority with no 'followmeid' in followme.conf
ASTERISK-09037: SIP SLA trunk fails codec negotiation after recent change
ASTERISK-09038: same extensions for different companies
ASTERISK-09039: [patch] Only apply externip on SIP, not on media streams
ASTERISK-09040: ChangeMonitor AMI command doesn't change the filename, and causes other problems
ASTERISK-09041: process_sdp: Insufficient information for SDP (m = '', c = '')
ASTERISK-09042: Asterisk segfaults upon receipt of a certain SIP packet (SIP Response code 0)
ASTERISK-09043: Improper response on receiving duplicated packets
ASTERISK-09044: [patch] IAXVARS does not work for authenticated IAX2 channels
ASTERISK-09045: AEL security risk in switch blocks
ASTERISK-09046: Asterisk crash in codec_alaw.c
ASTERISK-09047: 'Got SIP response 400 "Bad Request" back from 192.168.1.206' where calling a Thomson 2030
ASTERISK-09048: One-way audio on Polycom 501 after upgrade from 1.4.0 to 1.4.1
ASTERISK-09049: Make logging Unique ID a config option, rather than a compile option
ASTERISK-09050: [patch] SDP Header with invalid ip address in secondary "=c" line crashes Asterisk
ASTERISK-09051: Asterisk 1.4.0 Core Dump on voicemailbox consultation with g729 codecs but ok with g711ulaw codecs
ASTERISK-09052: Three way conferencing
ASTERISK-09053: ChanSpy and ExtenSpy don't get next channel
ASTERISK-09054: After some time Sip show peers cut 1 symbol (first)
ASTERISK-09055: [patch] unregister a peer via CLI command
ASTERISK-09056: Manager Dropping Events Under Moderate Call Load
ASTERISK-09057: Event: keep-alive header returns a 489 Bad Event response
ASTERISK-09058: Garbled audio using speex
ASTERISK-09059: variables.txt missing
ASTERISK-09060: [patch] counter for database show
ASTERISK-09061: Ekiga DTMF doesn't work in Asterisk 1.4.1 but works in 1.2.4-BRIstuffe
ASTERISK-09062: Sip channels != core channels
ASTERISK-09063: High latency (satellite) hangup on internal calls
ASTERISK-09064: [patch] email messages containing voicemail are not encoded properly on Exchange
ASTERISK-09065: h extension not executed on sip call ending
ASTERISK-09066: [patch] Attended transfer fails when hanging up before answer.
ASTERISK-09067: Asterisk 1.4.0: crash in put_unaligned_uint32
ASTERISK-09068: astxs is broken
ASTERISK-09069: Can't use gui if no writing Authorization
ASTERISK-09070: MOH does not follow language settings
ASTERISK-09071: same extensions for different companies
ASTERISK-09072: Festival hangs up a channel instead of doing its job
ASTERISK-09073: Problems
ASTERISK-09074: Sharing ODBC voicemail amongst several Asterisk servers
ASTERISK-09075: chanspy in private whisper by default
ASTERISK-09076: Fast-fingered DTMFs not working with IAX in the middle
ASTERISK-09077: Syntax and visual debug
ASTERISK-09078: TDM400P modules do not recognize lack of cable/dial tone
ASTERISK-09079: [patch] Improper locking in meetme results in a segfault
ASTERISK-09080: [patch] Counter for sip show registry
ASTERISK-09081: no video playing while announce
ASTERISK-09082: member "inuse" but get a ring
ASTERISK-09083: Early B3 not working properly in case of a cause code != 16 being returned from the remote party (calling from SIP to mISDN)
ASTERISK-09084: T38 outbound
ASTERISK-09085: Choppy MOH from SIP trunk
ASTERISK-09086: modules.txt explaining the old module format.
ASTERISK-09087: Asterisk doesn't send predefined Hangup causecode to sip_channel.
ASTERISK-09088: pri_hangup core dump
ASTERISK-09089: [patch] Voicemail may not play instructions for leaving a message
ASTERISK-09090: Message "The previous reload command didn't finish yet" when I make a "reload" from the CLI
ASTERISK-09091: "overlap dialing" overwrites exten
ASTERISK-09092: When "restart now" is issued Asterisk quits instead (started with -q) option
ASTERISK-09093: Error when trying to change password of a user by the voicemaila aplicacion
ASTERISK-09094: Error when trying to change password of a user by the voicemaila aplicacion
ASTERISK-09095: cisco enable password
ASTERISK-09096: cannot compile with DONT_OPTMIZE
ASTERISK-09097: [patch] Move AUDIORTPQOS to dialplan function
ASTERISK-09098: Manager causes segfault in generic_http_callback when run_tool run from the Gui.
ASTERISK-09099: ExtensionState action reports wrong extension state in 1.4.2
ASTERISK-09100: A segmentation fault occurs when two context have the same name
ASTERISK-09101: [patch] New function to change penalty value of current queue members
ASTERISK-09102: file format for h261/h263p
ASTERISK-09103: transfer=mediaonly : can't hear voice (machines all in the same net)
ASTERISK-09104: Offline G729 CODECS register
ASTERISK-09105: Unable to add calling rules after installing System Updates
ASTERISK-09106: all queues disappeared
ASTERISK-09107: menuselect compilation failure on Solaris 10/gcc-4.1.1
ASTERISK-09108: Inband DTMF Double digits being sent.
ASTERISK-09109: horrible audio with chan_cellphone
ASTERISK-09110: [patch] Valid RTP stream can be treated as invalid
ASTERISK-09111: chan_cellphone and bluetooth
ASTERISK-09112: Status-line in sipfrag notify is missing a CRLF
ASTERISK-09113: app_queue offers call then grabs it back with a timeout
ASTERISK-09114: [patch] res_config_odbc never reconnects
ASTERISK-09115: core show file formats
ASTERISK-09116: Allow calling MySQL 5 stored procedures from dialplan
ASTERISK-09117: Find_peer finds the first registered peer, resulting in inproper SIP channel display in the CDR and LOG
ASTERISK-09118: SIP Option rel100 not supported
ASTERISK-09119: Asterisk (chan_sip) hangs up call immediately after remote party picks up.
ASTERISK-09120: [patch] app_queue should use membername for COMPLETECALLER instead of channel ID
ASTERISK-09121: allowing IP
ASTERISK-09122: Call is hungup when SDP is not found in an ACK to 407
ASTERISK-09123: "sed -r" in mkpkgconfig not valid on Darwin
ASTERISK-09124: Misdn instability, ports go up/down when calls are passed
ASTERISK-09125: [patch] DTMF Is not relayed chan_sip[others]<->chan_gtalk <- ... -> chan_gtalk <-> chan_sip[others]
ASTERISK-09126: T.38 passthrough fails if caller offers T.38 in the initial INVITE
ASTERISK-09127: segmentation fault in find_callno using iax modem
ASTERISK-09128: unlock error in compare_weight
ASTERISK-09129: Authentication Failed when Logging in.
ASTERISK-09130: core dump in changethread
ASTERISK-09131: [patch] allow a call-specific setting of 'mohclass'
ASTERISK-09132: error response on INFO kills call
ASTERISK-09133: Response Timeout Deprecated
ASTERISK-09134: static-http resides under datadir
ASTERISK-09135: voicemails are not sent to users when the review option is chosen
ASTERISK-09136: Asterisk will not compile on FC5 due to errors compiling codec_zap.c
ASTERISK-09137: [patch] zaptel.init 1.2 and 1.4 to support zaphpec_enable
ASTERISK-09138: [patch] app_rtsp - playback RTSP media resources
ASTERISK-09139: cdr_odbc causes crash if database restarts
ASTERISK-09140: [patch] Configured codec packetization ignored
ASTERISK-09141: I want to send patch say.c for support Thai Language (Thailand)
ASTERISK-09142: [patch] Added a talk request for user in mute
ASTERISK-09143: Global variables defined in extensions.ael not reloaded when doing a reload, other parts of dialplan are.
ASTERISK-09144: channel.c: Bridge stops bridging channels .........<ZOMBIE>
ASTERISK-09145: Dynamic queue members marked invalid after asterisk is restarted
ASTERISK-09146: pbx_substitute_variables_helper_full logic (missing '}'
ASTERISK-09147: [patch] wrong DPC when communicating to multiple DPCs
ASTERISK-09148: [PATCH] Changes cdr to allow storage of up to 5 userfields
ASTERISK-09149: [PATCH] Send a correct version number for Manager Connection
ASTERISK-09150: [patch] RTP Payload 102 used by google for iLBC (chan_gtalk)
ASTERISK-09151: res_odbc.c environment variables setting
ASTERISK-09152: Modifying channel number produces mis-behaviour with HOLD or PARKing
ASTERISK-09153: zaptel compilation fails in wcusb.c on FC5 Stock kernel 2.6.15-1.2054_FC5
ASTERISK-09154: Loud noise heard after 2nd person joining conference bridge is announced.
ASTERISK-09155: Modify connection: Response 491 not handled according to RFC3261
ASTERISK-09156: [patch] res_config_mysql.c fails to compile
ASTERISK-09157: [patch] stop 100 trying on register
ASTERISK-09158: [patch] Error condition checking when connection to mysql is lost
ASTERISK-09159: [patch] "continue" doesn't perform the increment in "for" loops
ASTERISK-09160: sip show registry truncates the first character of the hostname in the last entry
ASTERISK-09161: Frequent crash with similar core on 3 different servers
ASTERISK-09162: Meetme can't use B410P's as timing source
ASTERISK-09163: [patch] parse_uri function chokes on some URIs
ASTERISK-09164: Missing warning about unknown parameters in sip.conf => No warning about misspelled "
ASTERISK-09165: Caller Id not getting passed
ASTERISK-09166: Voicemail
ASTERISK-09167: Channel DTMF stops taking new DTMF input. (Null Frame loop)
ASTERISK-09168: [patch] 2 same prototype
ASTERISK-09169: [patch] Problems with transcoding early media after "302 Moved temporarily"
ASTERISK-09170: [patch] Invite with codecs not supported by the caller in Early Bridge
ASTERISK-09171: Incoming early media problem after Asterisk generates its own early media
ASTERISK-09172: "request sent" hang on sip show regitry...
ASTERISK-09173: "Dial(mISDN/g:te/${EXTEN})" don't set up L2Link but "Dial(mISDN/1/${EXTEN}" works
ASTERISK-09174: improving zaptel's options documentation in menuselect
ASTERISK-09175: Videosupport messes up sip routing
ASTERISK-09176: AEL doesn't load default context at first time
ASTERISK-09177: [patch] Asterisk dont log when QueueMember is Added or Removed
ASTERISK-09178: "420 Bad extension" response is missing "Content-Length" field
ASTERISK-09179: unknown options reported in sip.conf in build_peer()
ASTERISK-09180: [patch] RTP Session id based on PID instead of network time stamp as suggested by RFC
ASTERISK-09181: Regression: Unable to Answer channels via Manager in *>1.2.16
ASTERISK-09182: Persistent CDR Start Disagreement
ASTERISK-09183: Hints problems with SLA
ASTERISK-09184: IAX doesn't automatically re-register after losing registration
ASTERISK-09185: Add ability to forward multiple users the same voicemail.
ASTERISK-09186: Asterisk 1.2.17 crashes with SIP message could not be handled error
ASTERISK-09187: segfault with core
ASTERISK-09188: Totally garbled sound when using speex codec
ASTERISK-09189: [patch] Evil nasty to allow CUT() to work in dundi.conf
ASTERISK-09190: When using CUT() in a DUNDi mapping, you always get a NULL value back
ASTERISK-09191: GROUP_COUNT can cause a crash if doing strange things with DUNDi mappings
ASTERISK-09192: mkdir '/var/spool/asterisk/voicemail/default//' failed
ASTERISK-09193: Unable to select service provider
ASTERISK-09194: Service providers not being recorded correctly
ASTERISK-09195: ast_register_application and loader misalignment
ASTERISK-09196: crash situation in chan_sip
ASTERISK-09197: wrong module error message when using voicemail in realtime mode
ASTERISK-09198: [patch] configurable voicemail playback controls
ASTERISK-09199: Proper escaping of JSON elements in GetConfigJSON
ASTERISK-09200: RTCP stats not being logged
ASTERISK-09201: [patch] improving asterisk's general labels in menuselect
ASTERISK-09202: Call Drops after 5 to 20 minutes
ASTERISK-09203: 'make distclean' and 'make clean' no longer work in Zaptel 1.4
ASTERISK-09204: Bad reinvite when called close the call...
ASTERISK-09205: changes to sla.conf are not picked up on asterisk reload
ASTERISK-09206: Calling VoiceMailMain within a macro with an SLA-style setup crashes asterisk
ASTERISK-09207: [PATCH] SIP Redirect through Transfer() app does not work properly
ASTERISK-09208: [feature request] CURL function has no apparent timeout on unreachable host
ASTERISK-09209: Queue uses minimum 4 channels, 2 calls, can not reinvite calls. possible g729 passthru violation.
ASTERISK-09210: memory leak in get_filestream
ASTERISK-09211: asterisk doesn't stop media session when sending BYE
ASTERISK-09212: There is no way to hide User-Agent field from SIP headers
ASTERISK-09213: DTMF tones not recognized or doubled on FXS port
ASTERISK-09214: [patch] DESTDIR should not be part of path names
ASTERISK-09215: Directed Pickup won't grab group calls - no target channel found
ASTERISK-09216: [patch] Asterisk sends wrong CSEQ in CANCEL if using INFO dtmf
ASTERISK-09217: [patch] Hungarian say.conf digits
ASTERISK-09218: Variables used in MixMonitor [command] section don't update after a dial command has been run.
ASTERISK-09219: app_queue with MixMonitor does not honor MONITOR_EXEC unless MONITOR_OPTIONS is set
ASTERISK-09220: SLA continue to ring when option ringdelay is set in sla.conf
ASTERISK-09221: username non alphanumerique username support
ASTERISK-09222: many WARNING messages while parsing users.conf
ASTERISK-09223: console dial can cause crash
ASTERISK-09224: [patch] h323 fails to build in 1.2: no target opt (as of 58008)
ASTERISK-09225: [patch] Fix Macro stack overflows permanently
ASTERISK-09226: [patch] separate sections in zapata.conf
ASTERISK-09227: MOH plays when a call is put on hold but will not stop when the call is picked back up
ASTERISK-09228: Crash in chan_local
ASTERISK-09229: problem with CDR record when using AUTOMON fetaure or res_monitor on outgoing calls (phone->*->telco)
ASTERISK-09230: MATH cannot be used with first parameter negative
ASTERISK-09231: ast_expr2.y fails on negative real numbers
ASTERISK-09232: Explicit Call Acceptance
ASTERISK-09233: [patch] for getting IE to accept SVG documents.
ASTERISK-09234: Q931 inband information not interpreted as alerting signal
ASTERISK-09235: GUI - File Editor
ASTERISK-09236: Enable moh for SLA trunk, like m option of dial command
ASTERISK-09237: announce-round-seconds > 30 doesn't make sense
ASTERISK-09238: host parameter for sip peers isn't updated back to "dynamic" when using an IP address for it
ASTERISK-09239: SIP_HEADER function after re-invite doesn't report headers in initial INVITE
ASTERISK-09240: [patch] same prototype
ASTERISK-09241: Segmentation fault when loading an empty res_config_mysql.conf
ASTERISK-09242: Asterisk 1.4.2 crash in put_unaligned_uint32
ASTERISK-09243: realtime prune (and others) may segfault asterisk (timing issue)
ASTERISK-09244: Polycom phones calling out on a TDM04B have a long delay before the called party can hear audio
ASTERISK-09245: fax - problem with t38, from linksys 2102 to cisco 5300
ASTERISK-09246: Asterisknow vmware-image version beta 5 includes an image asterisk-0.9.5-x86
ASTERISK-09247: Music on hold played between custom greeting and vm-intro.
ASTERISK-09248: [patch] Tone Zone Support for Philippines
ASTERISK-09249: G729 codec lock up when build with Jitter buffer
ASTERISK-09250: coredump in 1.2.16 in app_voicemail.c:5565
ASTERISK-09251: Asterisk deadlock
ASTERISK-09252: Asterisk segfaults when loading cdr_addon_mysql.so
ASTERISK-09253: asterisk crash when there's no space left on a local device or when there are too many extensions to handle?
ASTERISK-09254: call files with video phone no video
ASTERISK-09255: make install doesn't copy tools/zapscan.bin in /sbin
ASTERISK-09256: GUI zaptel conf => context could be choosen in a list
ASTERISK-09257: Asterisk crash
ASTERISK-09258: [patch] Incoming SIP Contact header truncated at 250 chars
ASTERISK-09259: spurious DNS lookups / SRV record lookups
ASTERISK-09260: AGI say number does not take gender argument
ASTERISK-09261: SIP - Bridge Call - Not all DTMF tones send when multiple digits are needed.
ASTERISK-09262: Asterisk now Beta 5 Crashes on core 2 Duo machine
ASTERISK-09263: [patch] Asterisk support of 802.1p priority marking
ASTERISK-09264: res_config_mysql segfaults after merging with 5881
ASTERISK-09265: Asterisk 1.4.2 crash ast_string_field_free_pools
ASTERISK-09266: no dialing on E1 possible
ASTERISK-09267: RTCP Statistics Broken for Asterisk-to-Asterisk calls
ASTERISK-09268: Asterisk crashes on Illegal instruction loading module codec_gsm.so
ASTERISK-09269: autocomplete for 'sip unregister' CLI command.
ASTERISK-09270: SIP dtmf signaling failing due to bad INVITE
ASTERISK-09271: Core Dump
ASTERISK-09272: [patch] config.h isnt deleted on a make distclean
ASTERISK-09273: Asterisk crashes with error "Unable to masquerade"
ASTERISK-09274: g729 register util fails if ethernet interfaces have been renamed
ASTERISK-09275: Queue does not fully allow '*' as exit digit
ASTERISK-09276: Backup fails during NFS mount
ASTERISK-09277: Unable to complete registeration / accept calls
ASTERISK-09278: Functionality to view pending asterisk console commands like "restart when convenient"
ASTERISK-09279: CALLERID(num) Unavailable Via GetVar
ASTERISK-09280: Does not compile chan_vpb.cc
ASTERISK-09281: Installation fails
ASTERISK-09282: agents assigned to multiple queues and weight=0 cause callers be handled in wrong order
ASTERISK-09283: The call through SIP provider remains ringing after hangup
ASTERISK-09284: Asterisk is crashing
ASTERISK-09285: calls out to Nortel CS1000 hang up
ASTERISK-09286: Unparked caller has ability to transfer
ASTERISK-09287: 603 declined do not stop after CANCEL in blind xfer
ASTERISK-09288: Notify sent to a non-existent call
ASTERISK-09289: updateconfig adding '= >value' instead of '=> value'
ASTERISK-09290: IAX2 crash
ASTERISK-09291: Notice and warnings of duplicate CDR process
ASTERISK-09292: Reload is heavily delayed and CLI gets unresponsive if res_snmp is enabled
ASTERISK-09293: Initial GUI buttons / Java Script Crash upon initial setup.
ASTERISK-09294: updatecdr in agents.conf is not working
ASTERISK-09295: function find_call in chan_sip.c
ASTERISK-09296: cna_modem thread freeze in restart_monitor() function.
ASTERISK-09297: Voicemail greeting message doesn't work with options
ASTERISK-09298: [patch] Allow execution of commands after app_dictate exits
ASTERISK-09299: ability to propogate NOTIFY request to the phone
ASTERISK-09300: Asterisk dies unexpectedly, cannot find out reason why. I suspect due to nagios monitoring on management interface.
ASTERISK-09301: Conference is not deleted on rename
ASTERISK-09302: Yet another Asterisk/Enum related bug
ASTERISK-09303: Asterisk GUI Behind Firewall
ASTERISK-09304: asterisk 1.4.2 crashes randomly a few times everyday
ASTERISK-09305: asterisk-core-sounds-*-1.4.6.tar.gz contain weird "x" file
ASTERISK-09306: Overlapping sounds in asterisk-core-sounds-* and asterisk-extra-sounds-*
ASTERISK-09307: Asterisk large number of processes problem
ASTERISK-09308: Code doesn't seem to use ani
ASTERISK-09309: [patch] Some PRI disconnect codes do not hangup correctly
ASTERISK-09310: MeetMe unable to write frame warning messages.
ASTERISK-09311: Memory leak problem in 'inboxcount' function when using IMAP/Realtime for Voicemail
ASTERISK-09312: MAX_INCLUDE_LEVEL not checked properly
ASTERISK-09313: ODBC produces Segmentation Fault
ASTERISK-09314: Core Dump - Asterisk Crashed
ASTERISK-09315: can't dial/make calls from skinny phone
ASTERISK-09316: Only delivery calls to Not in use users.
ASTERISK-09317: Sip registration on port different then 5060 fails Asterisk 1.2xx CentOS
ASTERISK-09318: users.conf hash323 option builds user not friend
ASTERISK-09319: iax2 calls are dropped when entering a MeetMe conference room
ASTERISK-09320: RTCP reports incorrect packet RTT (round trip time) statistics
ASTERISK-09321: [patch] strcasecmp in app_macro related to GOSUB returns a NULL causing a segfault.
ASTERISK-09322: Dump Manager Events to A Logfile
ASTERISK-09323: [patch] Making the absolute minimum time between position announcements configurable
ASTERISK-09324: Read() stops dial plan when file not found
ASTERISK-09325: Changing the name of a trunk in it's advanced propperties breaks the trunks config
ASTERISK-09326: App Read() Should not Disconnect on failed File
ASTERISK-09327: Debug logging not under if (option_debug)
ASTERISK-09328: Enhance app_meetme RealTime capabilities
ASTERISK-09329: RTPAUDIOQOS does not always report values
ASTERISK-09330: SIP SUBSCRIBE in LAN environment does not work because dialog is destroyed too quickly
ASTERISK-09331: I believe this fixes a BAD BAD BAD Scenario
ASTERISK-09332: Asterisk 1.4.3 crashes on startup
ASTERISK-09333: "Mute" IAX2 Extensions
ASTERISK-09334: DTMF Tones Get Caught In A Loop
ASTERISK-09335: DTMF tones get stuck on when passed through agent channel which is running on a zap channel
ASTERISK-09336: 1.4.2 randomly segfaults
ASTERISK-09337: When Using Multiple Queues, Agent Status Gets Stuck As "Busy" In Select Queues
ASTERISK-09338: CHANNEL function causes segfault
ASTERISK-09339: CHANNEL function - additional functionality request
ASTERISK-09340: tell browsers not to cache non-static content
ASTERISK-09341: CHANNEL QOS functions do not report stats after hangup
ASTERISK-09342: AEL gives syntax error on @ in a switch
ASTERISK-09343: Incoming DTMF signal plays continuously
ASTERISK-09344: Add support for Redial, Speeddials and Message button
ASTERISK-09345: [patch] language param doesn't work in zapata.conf
ASTERISK-09346: calling crash asterisk sometime
ASTERISK-09347: Cpusage
ASTERISK-09348: app_meetme unable to mix VoIP channels correctly
ASTERISK-09349: obsolete jitterbuffer config options in iax.conf.sample
ASTERISK-09350: improve jitterbuffer (jbtargetextra/minjitterbuffer)
ASTERISK-09351: Third-party modules that use last spandsp versions can't be build
ASTERISK-09352: Looking for non-existent workqueue.h in 2.4.18-19.7.x
ASTERISK-09353: Zaptel 1.2.17.1 can't produce LKM on 2.4.18-19.7.x
ASTERISK-09354: Dialplan application to disable/enable echo cancellation "on the fly"
ASTERISK-09355: Add MIB object astNumChanBridge to res_snmp
ASTERISK-09356: [patch] Close information leakage when passing a non-NULL terminated text frame
ASTERISK-09357: show channels CLI command only returns part of the full channel ID
ASTERISK-09358: Asterisk fails to compile in app_amd
ASTERISK-09359: Allow announcement message to be overridden by channel varible....
ASTERISK-09360: Segfault in chan_iax2.c - similar to 9278 and 9294
ASTERISK-09361: make install fails on: cannot stat `.libs/libchan_h323.so.1.0.1'
ASTERISK-09362: Under 1.4.4 the callerid(num) variable outputs the sip uri not the callerid num
ASTERISK-09363: Friendly Paths
ASTERISK-09364: service provider form blank in setup wizard
ASTERISK-09365: Images with providers not being shown correctly
ASTERISK-09366: BYE calls too fast when connected to a mediatrix box makes asterisk acts weird
ASTERISK-09367: No ringing heard on unattanded transfer
ASTERISK-09368: CallerID Name does NOT get passed correctly over PRI channel
ASTERISK-09369: Asterisk random segfault
ASTERISK-09370: SIT tones are not detected on PRI when dialing out numbers.
ASTERISK-09371: Asterisk CLI Busy Loop
ASTERISK-09372: Starting Musiconhold causes CANCEL
ASTERISK-09373: chan_misdn segfaults on a reload if there are no configured MISDN ports
ASTERISK-09374: Group function in Asterisk 1.4.4 fails on Local channels, it didn't in Asterisk 1.4.3
ASTERISK-09375: Show hints is not reliable at startup
ASTERISK-09376: rawman - command output should be stripped off the colors
ASTERISK-09377: [patch]Asterisk can't establish dialtone after brief hangup
ASTERISK-09378: atxfer target receives broken From SIP heaer
ASTERISK-09379: asterisk -rx does not work
ASTERISK-09380: chan_iax2.c deadlock in connection with iax2_destroy
ASTERISK-09381: make clean after make menuselect leaves tree unusable in 1.4.2.1
ASTERISK-09382: port or outboundproxyport not taken in account
ASTERISK-09383: Race condition leading to crash in chan_iax2
ASTERISK-09384: [patch] Random crash when checking version attribute in an incoming XML caps element
ASTERISK-09385: devicestate.c:__ast_device_state_changed_literal can't cope with iax (and sip?) peer names containing "-"
ASTERISK-09386: MeetMe members stay forever if softphone killed
ASTERISK-09387: When booting on kernel 2.6.19.7-0.4.gcc3.4.x86.i686 zaptel module not being loaded
ASTERISK-09388: [patch] MeetMeChannelAdmin application
ASTERISK-09389: sip channels get stucked
ASTERISK-09390: I upgraded for 1.2.12 to 1.4..4 and found that features.conf aplications no longer work
ASTERISK-09391: Segfault when bridge channel on iax_answer
ASTERISK-09392: [patch] acinclude.m4 crosscompilation macro's for pwlib/openh323
ASTERISK-09393: No res_pgsql.conf.sample and sql script
ASTERISK-09394: after a while, iax peers becomes unreachable/reachable
ASTERISK-09395: check_auth fails with 'username mismatch' when using multiple registrations to same host
ASTERISK-09396: Sending DTMF to agentcallbacklogin broken
ASTERISK-09397: Can not update system
ASTERISK-09398: asterisk destroy SIP dialog before auth BUY
ASTERISK-09399: Queue stops ring to members
ASTERISK-09400: chan_oss & chan_alsa in conference with zap channels
ASTERISK-09401: asterisk 1.2 from svn ... lock on shutdown
ASTERISK-09402: chan_h323 from svn branch 1.2 will not compile
ASTERISK-09403: chan_h323 compilation from SVN Branch 1.4
ASTERISK-09404: Asterisk Segfault under moderate call volume (appears to be crashing in logging call)
ASTERISK-09405: rtptimeout parameter non-functional on sip-to-sip
ASTERISK-09406: asterisk 1.2 from svn ... lock on shutdown
ASTERISK-09407: Regular expression parsing for ENUM entries on ENUMLOOKUP fails
ASTERISK-09408: Chan local with /n cause core dump
ASTERISK-09409: Crash when call parking after attended transfer
ASTERISK-09410: DTMF generation out of nowhere
ASTERISK-09411: Can't assign a Local channel as a static queue member with /n
ASTERISK-09412: Asterisk GUI setup wizard blank
ASTERISK-09413: BRIDGEPEER does not properly represent peer
ASTERISK-09414: problem in backport of chanspy whisper
ASTERISK-09415: Inconsistent periodic announcements when strategy != "ring all"
ASTERISK-09416: zero billsec in mysql db
ASTERISK-09417: [patch] KB1EC Timed Agressive Cancellation
ASTERISK-09418: jerky all calls until asterisk restart, probably jitterbuffer issue
ASTERISK-09419: I need restart asterisk because a channel Local is in Agent
ASTERISK-09420: Asterisk GUI setup wizard outbound calling rules error
ASTERISK-09421: Hint priority does not work with realtime
ASTERISK-09422: Can't pickup call parked on named (by ${PARKINGEXTEN}) parking slot
ASTERISK-09423: chan_iax2 crash when seeding dynamic peers
ASTERISK-09424: After instalation of "AsteriskNow beta 5" kernel does not boot
ASTERISK-09425: [patch] More than one group@category is allowed
ASTERISK-09426: Crash on Channel Hangup
ASTERISK-09427: [feature] autosystemname in asterisk.conf
ASTERISK-09428: Asterisk incorrectly parses the filenames in the outgoing directory and then stops monitoring /var/spool/asterisk/outgoing
ASTERISK-09429: [patch] extra spaces in configure script
ASTERISK-09430: [patch] doc/enum.txt no longer exists
ASTERISK-09431: [patch] Disposition still set to FAIL when it should be NO ANSWER or BUSY when dialling multiple SIP peers
ASTERISK-09432: cannot find a required config file (contactinfo.conf)
ASTERISK-09433: No rinback sent to outside caller while SLA Stations are ringing
ASTERISK-09434: Calls to SLA Stations don't disconnect.
ASTERISK-09435: Dial command always bridge calls directly when a call transfer is requested
ASTERISK-09436: Incorrect handling of native PLC frames in codec_speex.c
ASTERISK-09437: Attempt to park call to already taken parking slot result in weird behavior and requires asterisk restart
ASTERISK-09438: SIP Transfers To Parking Lot From Grandstream GXP-2000 Locks Up SIP Channel
ASTERISK-09439: The new compilation method is not explained in /README.
ASTERISK-09440: pre-dial digit collection
ASTERISK-09441: better logging for ACL deny
ASTERISK-09442: browsing to negative messages allowed after delete, folder change
ASTERISK-09443: Crash with duni call toi IAX channel
ASTERISK-09444: Realtime incorrect dialplan extension checking
ASTERISK-09445: BYE Authorization fails
ASTERISK-09446: With odbc realtime use for mailbox, crash when loss bbdd connection
ASTERISK-09447: make update and make uninstall
ASTERISK-09448: m Option ceases MOH early
ASTERISK-09449: Asterisk crashes with reload command
ASTERISK-09450: Asterisk segfaults if a message/name is recorded on voicemail with imap storage enabled.
ASTERISK-09451: [patch] "fixing" the saying of queue holdtime (for less than 2 minutes)
ASTERISK-09452: Using IAX and speex codec I get a crash with: Out of buffer space
ASTERISK-09453: [patch] unloading res_jabber causes asterisk to segfault
ASTERISK-09454: [patch] jabber username typo results in asterisk crash
ASTERISK-09455: authuser setting for imap storage does not work properly and prevents master user
ASTERISK-09456: Invalid BYE generated
ASTERISK-09457: zttranscode not unloaded by initd
ASTERISK-09458: Can I have a variable available after Dial() for the SIP Response?
ASTERISK-09459: Digital card support
ASTERISK-09460: Custom Calls and Simplify VoiceMenus-Queues-TimeBased Rules
ASTERISK-09461: asterisk crashes with chan_spy
ASTERISK-09462: RDNIS not properly decoded
ASTERISK-09463: T38MaxFaxBitrate interoperatibility
ASTERISK-09464: RSA peer auth broken in 1.4?
ASTERISK-09465: regcontext only works in realtime with rtcachefriends=yes
ASTERISK-09466: [branch] cdr_adaptive_odbc
ASTERISK-09467: [Patch] make uninstall
ASTERISK-09468: ./configure error
ASTERISK-09469: Zaptel stop function in init script causes kernel crash (on FC3)
ASTERISK-09470: Unresolved symbol ast_agi_register
ASTERISK-09471: Removing a step in a voice menu breaks numbering
ASTERISK-09472: when using RECORD FILE mp3, * is crashing
ASTERISK-09473: Button template for Cisco 7936 Conference station
ASTERISK-09474: T.38 Fax Outbound (revision of bug 9356)
ASTERISK-09475: Instalation Problems
ASTERISK-09476: Cannot dial using Clarent CPG 201, D-Link DG-104S works fine
ASTERISK-09477: [patch] Add support for regcontext and regexten to Skinny
ASTERISK-09478: can't login to GUI
ASTERISK-09479: SIP packets are shown in sip debug peer, but are not actually sent
ASTERISK-09480: reload command causes asterisk to crash
ASTERISK-09481: Crash using gtalk and jabber
ASTERISK-09482: Attempt to use 'Bridge' from manager in SVN-trunk-r61864 segfaults asterisk
ASTERISK-09483: Attempt to use 'Bridge' from manager in SVN-trunk-r61864 segfaults asterisk
ASTERISK-09484: Attempt to use 'Bridge' from manager in SVN-trunk-r61864 segfaults asterisk
ASTERISK-09485: Additional Call Setup Timeout
ASTERISK-09486: core dumped after 5 second as module load app_queue
ASTERISK-09487: Make "Incoming Calls" usable
ASTERISK-09488: trivial patch to use ast_strlen_zero() instead of strlen()
ASTERISK-09489: Set(CALLERID(num)=xxx) on SIP trunk not working, but on IAX works fine
ASTERISK-09490: Set(CALLERID(num)=xxx) on SIP trunk not working, but on IAX works fine
ASTERISK-09491: [patch] Wrapuptime persistent in a multiple queues with shared members enviroment.
ASTERISK-09492: SIP-phone -> SIP-PSTN-gateway gives silence with canreinvite=no
ASTERISK-09493: Add 12-hour clock support for skinny phones
ASTERISK-09494: Calls are terminated by the network when answered by IVR
ASTERISK-09495: warning negative timestamp with saydigits and festival
ASTERISK-09496: hints don't work
ASTERISK-09497: menuselect: configuration error: C compiler cannot create executables
ASTERISK-09498: menuselect: configuration error: C compiler cannot create executables
ASTERISK-09499: [patch] Add 'e' extension to handle application errors
ASTERISK-09500: [patch] Imap storage voicemail messages could not be played back from the phone except as gsm
ASTERISK-09501: [patch] Imap storage voicemail as gsm have duplicate attachments, whereas other formats have one.
ASTERISK-09502: Deadlock problem with agents, queues and libpri (stop accepting incoming calls in PRI lines)
ASTERISK-09503: Usage of weight option in queues.conf leads to deadlock
ASTERISK-09504: RINGING state not working with call-limit >0
ASTERISK-09505: [patch] SayUnixTime(,,S) (seconds) uses the us rules for saying numbers
ASTERISK-09506: Dial option S and reinvites
ASTERISK-09507: Asterisk does not verify state of the channel IAX
ASTERISK-09508: [patch] Call counter not updating for looping request
ASTERISK-09509: [patch] chan_gtalk simultaneous hangup null pointer
ASTERISK-09510: Kernel crashes when loading Zaptel with TE410P with HW echocan
ASTERISK-09511: Integrate Adhearsion into GUI
ASTERISK-09512: [patch] Multirow results
ASTERISK-09513: Zaptel init file issue with HPEC (1.2.17.1 tarball)
ASTERISK-09514: [patch] Datastore docs outdated
ASTERISK-09515: after call to not available sip-channel, hint stays InUse
ASTERISK-09516: Ring garbled on FXS when Dial(Zap/1,,d) and unknown extension pressed
ASTERISK-09517: Module unload method not called on shutdown
ASTERISK-09518: MACRO_EXTEN, MACRO_... is empty in AEL
ASTERISK-09519: usedistinctiveringdetection=yes doesn't set the variable in chan_zap.c
ASTERISK-09520: GotoIf doesn't seem to work in the h extension.
ASTERISK-09521: dring1 values aren't properly read into asterisk
ASTERISK-09522: distinctivering context isn't getting set
ASTERISK-09523: Asterisk [IMAP] crashes if a vm message is left, then vm is checked but phone hung up before pswd entered
ASTERISK-09524: [patch] Trivial check, don't try to expire an unregistered peer via CLI command 'sip unregister'
ASTERISK-09525: [patch] CLI unregister command 'iax2 unregister <peername>'.
ASTERISK-09526: [patch] queues-with-callback-members.tex for MACRO_EXTEN problem
ASTERISK-09527: [patch] app_minivm.c deadlock when no configuration found.
ASTERISK-09528: [patch] major fix for SIP video codec negociations
ASTERISK-09529: [patch] support for Cirpack keepalive packets and RFC 2388 DTMF inside SIP INFO packets
ASTERISK-09530: Asterisk segfaults with realtime dynamic engine mysql if macro is mapped
ASTERISK-09531: Wrong "Got 200 OK on REGISTER that isn't a register"
ASTERISK-09532: Missing soundfile: vm-duration
ASTERISK-09533: Access-URL sip header causes Nokia E series to freeze
ASTERISK-09534: constant core dumps with ~100 calls
ASTERISK-09535: Distinctive Ring Detection Doesn't Use Configured value.
ASTERISK-09536: File descriptors all messed up
ASTERISK-09537: asterisk crash few minutes after startup
ASTERISK-09538: Subscriptions don't work on sip created in users.conf
ASTERISK-09539: Asterisk core dumps on transfer with Snom phones
ASTERISK-09540: With realtime users Chan_sip makes causeless load on database server
ASTERISK-09541: Bug at reload_config() leads to segfault
ASTERISK-09542: Use of reserved C++ terms
ASTERISK-09543: iax2 prune realtime [peer] can't work
ASTERISK-09544: After QueuePause(true) and queue inboundcall is delivered to sip
ASTERISK-09545: Out of idle IAX2 threads for I/O, pausing
ASTERISK-09546: misdn channel active but it isn't true
ASTERISK-09547: Document TRANSFER_CONTEXT in extensions.conf.sample
ASTERISK-09548: Asterisk crashes random within chan_sip
ASTERISK-09549: regcontext only works in realtime with rtcachefriends=yes on all channels (SIP and IAX2)
ASTERISK-09550: Bye authorization working only one way.
ASTERISK-09551: cannot update asterisknow via gui or console
ASTERISK-09552: line no needed in the ringgroups
ASTERISK-09553: Queue Calls get hung
ASTERISK-09554: Small updates to the Doxygen Documentation
ASTERISK-09555: [patch] /etc/init.d/asterisk is not "Linux Standard Base" compatible
ASTERISK-09556: Subscriptions deleted after reload
ASTERISK-09557: Reload cause crash
ASTERISK-09558: [patch] app_minivm.c code cleanup and not needed LOCKs/frees
ASTERISK-09559: Transfer button on GXP200 cause channel deleting
ASTERISK-09560: When using a perl agi script STREAM FILE wont return escape digits
ASTERISK-09561: Calling macros from Dial(), MACRO_RESULT is impossible to populate properly due to extension handling.
ASTERISK-09562: [patch] asterisk man page (changes in the CLI commands)
ASTERISK-09563: When answering, line from CO continues to ring.
ASTERISK-09564: ast_channel_masquerade causes crash
ASTERISK-09565: Crash on successive reloads
ASTERISK-09566: [patch] comments in res_config_sqlite.c: it's called res_config_sqlite
ASTERISK-09567: Interoffice IAX2 trunk kills VoipStreet trunk
ASTERISK-09568: replacing of ast_exists_extension for ast_goto_if_exists
ASTERISK-09569: Notify from Sipura SIP fails with SIP/2.0 489 Bad event
ASTERISK-09570: [patch] Using AGI command "say number 0 *" results in no audio output.
ASTERISK-09571: [patch] Don't use malloc() & memset(), use ast_calloc().
ASTERISK-09572: [patch] changed 'show parkedcalls' to 'parkedcalls show'
ASTERISK-09573: [patch] mixmonitor CLI command (autocomplete command options)
ASTERISK-09574: Doxygen updates
ASTERISK-09575: [patch] autoconf libcurl detection when crosscompiling
ASTERISK-09576: [patch] C++ Compiler warning on ast_string_field_set
ASTERISK-09577: Asterisk restart with segmentation fault while IAX peer is registered
ASTERISK-09578: C++fication of channels/*.c
ASTERISK-09579: [patch] Don't use malloc() & memset(), instead use ast_calloc() and ast_malloc().
ASTERISK-09580: AseriskNOW Web UI Doen't Accept Passwords Beginning with Percent Sign
ASTERISK-09581: Progess tone not passed.
ASTERISK-09582: [patch] asterisk-addons crosscompilation fixes
ASTERISK-09583: Setting a variable in the AGI using a dial plan function.
ASTERISK-09584: Installation Fails
ASTERISK-09585: CLI in GUI does not work
ASTERISK-09586: segfault with voicemail and ODBC
ASTERISK-09587: Switch construct breaks the dialplan flow if it doesn't have a default case
ASTERISK-09588: extended safe_asterisk with log capabilityes
ASTERISK-09589: AEL macros do not allow "includes" construct
ASTERISK-09590: DISA + Queue Login + *# sequence fails with number validity
ASTERISK-09591: [patch] allowmultiplelogin variable not being initialized and default set to 'no'.
ASTERISK-09592: Doxygen Update res_jabber
ASTERISK-09593: Support for current generation of Cisco phones (79X1)
ASTERISK-09594: Sip doesn't issue re-invite when before bye when hangup is caused by timeout on channel
ASTERISK-09595: Deadlocks cause SIP clients to stop responding
ASTERISK-09596: [patch] 'agi debug off' is not working.
ASTERISK-09597: [patch] Add bitwise & and bitwise | functionality
ASTERISK-09598: Doxygen Updates res folder
ASTERISK-09599: [patch] small typo in chan_zap with SS7
ASTERISK-09600: g.729 codec on dell 2950 server
ASTERISK-09601: g.729 codec on dell 2950 server
ASTERISK-09602: [patch] Authentication support for SIP NOTIFY requests
ASTERISK-09603: translate.c displays false warning during startup
ASTERISK-09604: the GUI is missing some of the functions on the left panel
ASTERISK-09605: core dump with misdn in nt mode
ASTERISK-09606: Infinite loop on SIP registration
ASTERISK-09607: Asterisk segfaults with total randomness
ASTERISK-09608: Make a variable timeout on attended transfer (atxfer)
ASTERISK-09609: [patch] for debug output for ss7
ASTERISK-09610: Voicemail greeting accepts automatically
ASTERISK-09611: H and h parameters do not hang up when * is dialed
ASTERISK-09612: Entering the admin pin for MeetMe will log you in as an admin even if the 'a' option was not specified.
ASTERISK-09613: ex-girlfriend feature does not load properly
ASTERISK-09614: [patch] logfile name is not set from config file
ASTERISK-09615: When Asterisk is issued 2 Invites with the same CSeq it does not read the corrisponding ACKs
ASTERISK-09616: [patch] Persistent queues with Realtime
ASTERISK-09617: * is looping and CPU goes at 100%
ASTERISK-09618: No option for PRI when setting up GUI
ASTERISK-09619: Add Skinny support to GUI
ASTERISK-09620: chan_zap Core Dump - Incoming Call
ASTERISK-09621: Zaptel 1.4.2.1 Init scripts cause a kernel panic
ASTERISK-09622: parameter count check for realtime handler
ASTERISK-09623: IAX2 -> ZAP PRI doesn't connect end-to-end when receipient picks up. SIP -> ZAP works from same phone
ASTERISK-09624: Dundi and Dyndns-update
ASTERISK-09625: 1.4.4 sends Re-INVITE twice, resulting in code 491 "Request pending" and call termination by Asterisk
ASTERISK-09626: Asterisk segmentation fault
ASTERISK-09627: Responses to Manager Commands Should Be Called 'Responses' and not 'Events'
ASTERISK-09628: [patch] log_debug
ASTERISK-09629: janitor project - remove priority jumping in trunk
ASTERISK-09630: fskmodem and tdd lack documentation of proper use...
ASTERISK-09631: bug cdr_mysql: Out of memory error (insert fails)
ASTERISK-09632: [patch] Add autocomplete for command: 'manager show user'.
ASTERISK-09633: fskmodem and tdd lack documentation of proper use...
ASTERISK-09634: [patch] MySQL stored procedures - FULL SUPPORT
ASTERISK-09635: [patch] Makes displayconnects a per user config option.
ASTERISK-09636: New manager event class - cdr
ASTERISK-09637: Zap Channel Bridging
ASTERISK-09638: Set() works different in realtime
ASTERISK-09639: Wrong CLI command in sip.conf.sample
ASTERISK-09640: IAX peers becomes unreachable/reachable
ASTERISK-09641: GUI removes escape character backslash
ASTERISK-09642: [branch] Transfer implementation
ASTERISK-09643: Crash on reload
ASTERISK-09644: Inband is not working (commented out) for H323
ASTERISK-09645: Asterisk crash in ast_readframe
ASTERISK-09646: StartMusicOnHold application stops inband ringing on all extension
ASTERISK-09647: adding labels to realtime
ASTERISK-09648: CLI commands 'core show [audio|video|image] codecs' not working
ASTERISK-09649: Crash while updating hints
ASTERISK-09650: play_mailbox_owner uses wrong variable with ODBC enabled
ASTERISK-09651: AttributeError: InstallClass instance has no attribute 'hideEverything' on installition of version Beta5 1/2
ASTERISK-09652: 404 error
ASTERISK-09653: Peer status is alway unknown
ASTERISK-09654: dead lock
ASTERISK-09655: [patch] Makes CLI command 'Agent logoff' autocomplete only the logged in agents.
ASTERISK-09656: many unnecessary CDRs using Queue cmd
ASTERISK-09657: when using dial and "&" with zap groups, other lines stop ringing when a free zap line is found
ASTERISK-09658: Duplicate sound files inside the "core" and "extra" package
ASTERISK-09659: chan_iax2 reports dropping frame from (IP) due to frame already in process during calls
ASTERISK-09660: Janitor Project - ast_debug() conversion
ASTERISK-09661: Not clearing previously dialed number
ASTERISK-09662: Issues with DTMF not coming in on time
ASTERISK-09663: activate asterisk bjitterbuffer for mISDN
ASTERISK-09664: Calling mysql_free_result without a valid result set crashes asterisk
ASTERISK-09665: Asterisk is locked in ast_get_channel_by_name_locked call
ASTERISK-09666: Chinese Taiwan (TW) language support
ASTERISK-09667: Revision to Chinese Taiwan (TW) support of say.c
ASTERISK-09668: Error in Custom Installation of *NOW B5
ASTERISK-09669: GROUP_LIST() issue
ASTERISK-09670: inUse update on SIP trunk peer
ASTERISK-09671: [patch] Janitor project ast_strlen_zero().
ASTERISK-09672: [patch] AgentCallbackLogin deprecated so __login_exec callbackmode is unnecessary.
ASTERISK-09673: mixmonitor does not create missing monitor directories
ASTERISK-09674: [patch] Add AMI event generation on channel creation failure.
ASTERISK-09675: [branch] res_jabber over OpenSSL
ASTERISK-09676: Channel variable ${SLATRUNK_STATUS} returns SUCCESS on ringtimeout and on answered conditions.
ASTERISK-09677: Crash when restart requested on PRI span
ASTERISK-09678: Resuming held call fails from Broadsoft Call Manager.
ASTERISK-09679: Resuming held call fails from Call Manager
ASTERISK-09680: Transfers resulting in a different negotiated codec result in no voicepath
ASTERISK-09681: Session Audit fails when audit occurs after AsteriskNOW restart
ASTERISK-09682: DTMF fails when RFC2833 offered by not negotatiated
ASTERISK-09683: MWI (Message Waiting Indicator) Notify from BroadWorks to AsteriskNOW fails
ASTERISK-09684: [patch] Janitor project use ast_strdup() instead of strdup()
ASTERISK-09685: [patch] Janitor project convert to ast_debug()
ASTERISK-09686: Premature Channel Hangup on AGI
ASTERISK-09687: RTP Reinvite fails when more than 1 softswitch in place.
ASTERISK-09688: Asterisk crash while working - see gdb file
ASTERISK-09689: [patch] A few code simplifications
ASTERISK-09690: configure does not find ODBC or does not explain what it did not like about the ODBC
ASTERISK-09691: [patch] ast_mkdir in place of ast_safe_system("mkdir -p ...")
ASTERISK-09692: duplicate sound files in the "core 1.4.7" and "extra 1.4.6" sound packages
ASTERISK-09693: Hold sends invalid media to peer, thus rejected by peer and call disconnected (SVN 69583)
ASTERISK-09694: [patch] Converted cdr_manager to use custom cdr format
ASTERISK-09695: Menuselect won't consider local members as satisfied dependencies
ASTERISK-09696: exit from make should only happen on errors
ASTERISK-09697: Makefile Header
ASTERISK-09698: Timelimit for options L and S is set to the inbound channel not the outbound
ASTERISK-09699: recent change to safe_asterisk fails with some shells
ASTERISK-09700: runasuser/runasgroup used even for -r means reconnecting users must be root
ASTERISK-09701: [patch] libss7 add support national, international prefix and work with Nature of Address Indicator
ASTERISK-09702: ASTVARLIBDIR/ASTVARRUNDIR locations on *BSD
ASTERISK-09703: WebGUI forces change of password with the form not doing anything.
ASTERISK-09704: Asterisk 1.2.13 crashes after call forward to a local channel
ASTERISK-09705: attempting to read rtpqos w/o specifying type or field will segfault
ASTERISK-09706: mxml fails while trying to http query meetme
ASTERISK-09707: Asterisk 1.4.5 crash with asterisk-addons-1.4.1 using mysql
ASTERISK-09708: Queue member gets called when user hangs up during periodic announcement
ASTERISK-09709: "ast_bridge_call: Bridge failed" after ChannelRedirect
ASTERISK-09710: Asterisk eating 100% CPU
ASTERISK-09711: CDR channel column doesn't change to Agent/agent_id when using AgentMonitorOutgoing(c) with Originate script
ASTERISK-09712: Request alternate column name "call-limit" to "call_limit" for SIP realtime configuration.
ASTERISK-09713: app_stack.c causes asterisk crash
ASTERISK-09714: Update to CREDITS file so it can be parsed
ASTERISK-09715: Asterisk resets AMAflag if call being answered
ASTERISK-09716: crash on ast_rtcp_write
ASTERISK-09717: h323 crash on high volumn calls
ASTERISK-09718: Asterisk hang/stop working on console suspend
ASTERISK-09719: Memory leaks in cdr_pgsql
ASTERISK-09720: [patch] uri_decode changing + signs to spaces (' ')
ASTERISK-09721: Race condition on cdr_pgsql config reload
ASTERISK-09722: CDR(src) cannot be set independently of ANI
ASTERISK-09723: Adding ZRTP security protocol support for asterisk
ASTERISK-09724: Asterisk-1.4.5 does not compile with voicemail ODBC storage
ASTERISK-09725: Asterisk 1.2.18 crashes after "reload"
ASTERISK-09726: nothing happens on ring if phones RJ45 is disconnected should go to Voice Mail
ASTERISK-09727: backup and restore to different location
ASTERISK-09728: [patch] Calls to Google Talk failing at 2nd call.
ASTERISK-09729: Ringback stops after 183 Session Progress is received from a SIP UA that is part of a multi UA Dial attempt
ASTERISK-09730: DTMF Problem
ASTERISK-09731: outgoing speech problem
ASTERISK-09732: Agi exits without any possibility to catch on stream file
ASTERISK-09733: Missing Remote-retrieve on attended transfer
ASTERISK-09734: SIP Reinvite Packets incorrect Sequence causes no audio when more than 1 softswitch in callpath.
ASTERISK-09735: res_agi launch_script say: File does not exist
ASTERISK-09736: Asterisk does not parse n extensions correctly
ASTERISK-09737: random crashes in channel.c
ASTERISK-09738: [patch] chan_sip does not handle 300 Multiple Choice properly
ASTERISK-09739: crash in devicestate.c
ASTERISK-09740: DBPut fails on blank value
ASTERISK-09741: Typo in res/res_features.c causes segmentation fault
ASTERISK-09742: [patch] ast_debug
ASTERISK-09743: [patch] ast_indicate can be executed on NULL channel
ASTERISK-09744: Allow linking against system IMAP library
ASTERISK-09745: Misspelt "Name of the currentl channel." asterisk-mib.txt
ASTERISK-09746: Fixes for CLI auto complete
ASTERISK-09747: AST_FRAME_DTMF_BEGIN not triggered on SIP/IAX channels during READ command
ASTERISK-09748: Libss7 and Siemens EWSD (ITU style),SS7 link crashes.
ASTERISK-09749: NOTIFY race condition when state changes happen very fast
ASTERISK-09750: NULL voicemail state terminates Asterisk
ASTERISK-09751: Incorrect parsing of IMAP mailbox address when using "authuser" authentication
ASTERISK-09752: [patch] core show channels command from manager causes seg fault
ASTERISK-09753: Use more consistent CallerID naming in IMAP mail headers to fix "unknown caller" voicemail info.
ASTERISK-09754: [patch] Add additional call counter information to res_snmp
ASTERISK-09755: Asterisk rfc2833 DTMF fails with SIP service providers
ASTERISK-09756: Reverted patch for issue # 10010 (was: Asterisk eating 100% CPU)
ASTERISK-09757: MixMonitor does not work in MeetMe using Zap Channels
ASTERISK-09758: [patch] Create the directory if it does not exist.
ASTERISK-09759: crash after changing folder and saving message
ASTERISK-09760: Func_odbc causes crash
ASTERISK-09761: User in meetme once muted is unable to access menu while muted (and therefore can't unmute self)
ASTERISK-09762: RTCP NTP Skew Detected
ASTERISK-09763: CDR dst and dcontext field wrong information depending if caller or callee hangs up first
ASTERISK-09764: SIP compile warnings, cashes crash on invite received
ASTERISK-09765: asyncgoto_exec: ChannelRedirect failed for SIP/keshav
ASTERISK-09766: [patch]: Added control over the spying party and ability to spy on calls both ways, including incoming on queue.
ASTERISK-09767: [branch] Add milliseconds to CLI display and make the -T option work as advertised
ASTERISK-09768: with odbc storage enabled, hanging up while recording a greeting kills asterisk
ASTERISK-09769: MS SQL 2005 escape character problem
ASTERISK-09770: One-way audio (one-way perfect and one-way distorted) with SK65
ASTERISK-09771: Error on SIP channel lock / unlock with cdr_tds
ASTERISK-09772: IAX2 protocol flaw in IC_NEW could cause reflective amplification DoS
ASTERISK-09773: With Digium transcoder card G.723 conversations involving prompts from Asterisk are still not possible
ASTERISK-09774: Logging of IP address of originating call...
ASTERISK-09775: RTCP Read Too Short generate strange DTMF tones on call.
ASTERISK-09776: add support for wideband speex (Openwengo variant)
ASTERISK-09777: When i restart asterisk gracefully or when..... it will segfault.
ASTERISK-09778: Zap Channel stays in use when Hangupcause = 1
ASTERISK-09779: Application and function to determine existence of extensions but not jump to it yet
ASTERISK-09780: SayAlpha() and SayPhonetic() fail when encountering unknown symbols, hanging up on the calling channel.
ASTERISK-09781: Read application buffers input DTMF tones
ASTERISK-09782: [patch] convert to ast_debug() in asterisk-addons
ASTERISK-09783: addons rev 406 is not compilled, because ast_realloca removed
ASTERISK-09784: Calls get disconnected during a conversation
ASTERISK-09785: Call OffHold Event in Asterisk
ASTERISK-09786: not register
ASTERISK-09787: Unable to join queue bevor reload
ASTERISK-09788: "Maximum PBX stack exceeded" raised by including more than AST_PBX_MAX_STACK *not nested* contexts
ASTERISK-09789: [branch]Arrays Requested
ASTERISK-09790: [branch] Event Based CDR system -- CEL (channel Event Logging)
ASTERISK-09791: saying unixtime in 24Hrs format in french says two times "o'clock"
ASTERISK-09792: Requesting action=Agents throught http manager interface doesn't return correct talkingto values when agent is talking to SIP.
ASTERISK-09793: Asterisk Crashes with: ERROR[13628] /usr/src/asterisk-1.4.5/include/asterisk/lock.h: channel.c line 2522 (ast_indicate_data):
ASTERISK-09794: Update for some errors with doxygen
ASTERISK-09795: Compiler Error
ASTERISK-09796: RetryDial application doesn't accept 0 for retries
ASTERISK-09797: A permanent INVITE that gets BUSY a SIP user
ASTERISK-09798: [Realtime] Wrong Matching within extensions table
ASTERISK-09799: Ability to Pause/Resume voicemail during playback
ASTERISK-09800: Local channel remains after hangup if the call fails to start
ASTERISK-09801: Record-Route fields are not copied when the response is 18x
ASTERISK-09802: [patch] manager_event tweak for the AgentCalled event
ASTERISK-09803: configure failed using --with-mysqlclient=/usr/local/mysql
ASTERISK-09804: A ServicelevelPerf value is not initialized correctly.
ASTERISK-09805: Bad operator (OBJECT-IDENTIFIER): At line 739 in /usr/local/share/snmp/mibs/ASTERISK-MIB
ASTERISK-09806: Playback does not produce any sound in channel
ASTERISK-09807: DoS (as a crash or not) when the Call-Id: of successive SUBSCRIBE packets is identical + Authentication
ASTERISK-09808: The language setting for Voicemail in sip.conf is not applied when the call is done with an Originate (AMI)
ASTERISK-09809: non-stop "Event: MeetmeMute" sent to manager port when joining meetme
ASTERISK-09810: MusicOnHold() drops the channel if the specified class does not exist
ASTERISK-09811: asterisk crashes on reloading after bindaddr is changed
ASTERISK-09812: If 'retry' is set to 0, it gets treated as though it's set to 5.
ASTERISK-09813: Queue timeouts are considerably higher than expected
ASTERISK-09814: Module SRTP can't loaded
ASTERISK-09815: Module SRTP can't loaded
ASTERISK-09816: Module SRTP can't loaded
ASTERISK-09817: Module SRTP can't loaded
ASTERISK-09818: Asterisk Random Segfault (might coincide with parking a call)
ASTERISK-09819: Large memory leak in dns.c:ast_search_dns()
ASTERISK-09820: Crash on leaving message with IMAP storage
ASTERISK-09821: IMAP storage need strange configuration
ASTERISK-09822: changes ast_exists_extension to function ast_goto_if_exists
ASTERISK-09823: Can't dial out on chan_skinny
ASTERISK-09824: Incorrect destination mailbox when forwarding message with IMAP storage
ASTERISK-09825: When removing a class from musiconhold.conf and reloading the module, the class is not removed from memory
ASTERISK-09826: [patch] crash when decoding callerid
ASTERISK-09827: IAX2 hangs on a busy server
ASTERISK-09828: phone continues to ring after been picked up (handle_invite_replaces)
ASTERISK-09829: No license statemement in the sound tarballs
ASTERISK-09830: Asterisk wont compile anymore
ASTERISK-09831: if asterisk is received get perpetual dtmf
ASTERISK-09832: bootstrap.sh not working on OpenBSD
ASTERISK-09833: [patch] Fix for response to CoreSettingsAction
ASTERISK-09834: Add the ability to not play musiconhold but plain old ringing instead
ASTERISK-09835: autofill reports and behaves in 2 different ways
ASTERISK-09836: BUG in dialplan save
ASTERISK-09837: Service level might exceed 100%
ASTERISK-09838: login succeds even if the same agent is logged in via AgentCallbackLogin
ASTERISK-09839: In voicemail, the language of the displayed date inside the mail (VM_DATE) doesn't take into account the system's locale.
ASTERISK-09840: chan_sip does not use calling number in rpid for guest
ASTERISK-09841: chan_mobile no sound with iogear gbu221 (class2) dongle/samsung cellphone
ASTERISK-09842: Manhattan MII-794 class 2 usb/bluetooth adapter: asterisk crashes when call is bridged.
ASTERISK-09843: Incorrect notify handling when hints contain multiple devices
ASTERISK-09844: Sometimes my SIP extension gets on Hold
ASTERISK-09845: The device state of this queue member, SIP/XXXXX, is still 'Not in Use' when it probably should not be!
ASTERISK-09846: Possible injection attack in dialplan - time consuming to program around.
ASTERISK-09847: CLIR doesn't work on outgoing SIP calls
ASTERISK-09848: wrapuptime doesn't work for an agent logged on via AgentLogin
ASTERISK-09849: [patch] Typo in description
ASTERISK-09850: Asterisk crashes on GROUP_COUNT with a category
ASTERISK-09851: uninstall removes non-Asterisk stuff
ASTERISK-09852: Asterisk fails to install with kernel 2.6.21.5
ASTERISK-09853: 10075 fix in 1.4.7.1 breaks MS SQL 2000
ASTERISK-09854: autologoff, wrapuptime and ackcall don't persist
ASTERISK-09855: Change preprocessing commands in isdn_lib_intern.h
ASTERISK-09856: Agent logged on via AgentLogin isn't logged off by "agent logoff" CLI command with soft option
ASTERISK-09857: asterisk crashes when phone jack tries to generate dialtone
ASTERISK-09858: Music on hold stops on blind transfer.
ASTERISK-09859: [patch] Change logging to be more similar to system logging
ASTERISK-09860: Invalid memory reference crash in aji_handle_presence
ASTERISK-09861: [patch]: Added automixmonitor feature / ability to use mixmonitor to record queue conversations on demand.
ASTERISK-09862: RetryDial exits extension when attempting to "Retry" if announce file does not exist.
ASTERISK-09863: False error message from MOH at startup
ASTERISK-09864: Inbound inband DTMF broken passing on out of band (process_ast_dsp)
ASTERISK-09865: Asterisk does not send events when a monitor is started or stopped
ASTERISK-09866: New feature: Turn core debugging on for a file
ASTERISK-09867: [patch] Voicemail does not get copied to additional mailboxes when using ODBC storage
ASTERISK-09868: Voicemail does not attach audio file to email when sending to multiple voicemail boxes with ODBC storage
ASTERISK-09869: ChanSpy doesn't work fine for Zap channels from 1 to 9 however works fine for zap channels > 9
ASTERISK-09870: my asterisk comes down in flames randomly, it appears to be related to chanspy
ASTERISK-09871: Deadlock in chan_local causing a channel to be 'dummy' and impossible to soft hangup it.
ASTERISK-09872: Important vulnerability after native transfer: the transferred gets context privileges
ASTERISK-09873: [patch] Random replacement of channel name with other text in queue log entries
ASTERISK-09874: Users with the same last name are not all listed in the directory. Only one of them is.
ASTERISK-09875: Directory not reading from users.conf
ASTERISK-09876: idle console disconnects
ASTERISK-09877: Updates to .h doxygen
ASTERISK-09878: Encore Bluetooth Dongle: ENUBT-C1E (class1) No voice, crash on hangup
ASTERISK-09879: Results containing spaces from FETCH cause ast_yyerror()
ASTERISK-09880: [feature] [patch] Add ability to execute 'h' on the called peer in addition to the calling channel
ASTERISK-09881: Extraneous newline in AGI debugging output
ASTERISK-09882: sometimes asterisk crashes after reload with changes on agents or queues
ASTERISK-09883: ast_check_hangup does not return true after Hangup() has been called.
ASTERISK-09884: [patch] res_agi, some long outstanding hacks, move chan->_softhangup to ast_check_hangup
ASTERISK-09885: Peer configured in configuration files disapear from `sip show peers`
ASTERISK-09886: Asterisk reinitialize zaptel channels where send DTMF
ASTERISK-09887: rev71656 and rev71065 makes DeadAGI exit after channel hangup
ASTERISK-09888: Asterisk reinitialize zaptel channels
ASTERISK-09889: ACD Agent reported as busy when avalible
ASTERISK-09890: the right parentheses is in wrong position of function ast_test_flag
ASTERISK-09891: [patch] proper user information layer 1 handling
ASTERISK-09892: Segmentation fault at channel.c:3275
ASTERISK-09893: HTTP manager delivery invalid XML
ASTERISK-09894: chan_mobile: if not load kernel module l2cap, asterisk crash
ASTERISK-09895: When Sending A "restart when convenient" There Is a Segfault
ASTERISK-09896: undefined codecs in AST_FORMAT_LIST
ASTERISK-09897: compile errors and choppy sound with gcc 4.2.1
ASTERISK-09898: Incorrect handling of VNAK retransmission
ASTERISK-09899: Extension Status does not change to 8 durring ring
ASTERISK-09900: Sound problems when bridging sip channels with packetization level different than 20ms
ASTERISK-09901: [patch] where possible replace uses of channel->_softhangup with ast_check_hangup(channel)
ASTERISK-09902: chan_iax2 qualify marks peers unreachable when they are reachable
ASTERISK-09903: Incorrect handling of out of order full frames
ASTERISK-09904: polarity detection fixes (chan_zap)
ASTERISK-09905: chan_iax2 sometimes incorrectly destroys channel immediately after sending NEW and getting AUTHREQ
ASTERISK-09906: Support for 7921 Phones
ASTERISK-09907: [patch] Fix of crash on unloading chan_mobile.so
ASTERISK-09908: [patch] Improve to work with new say
ASTERISK-09909: Agi in live & dead channels? Don't drop DeadAGI from live.
ASTERISK-09910: Separation of filenames if the filename consist of many "."
ASTERISK-09911: This makes sure that on 'reload', auth list is cleared
ASTERISK-09912: chan_iax2 VNAK storm seen
ASTERISK-09913: Dont download sound files if they exist
ASTERISK-09914: [patch] allow extra flags for cross compilation
ASTERISK-09915: [patch] uClibc has no res_ninit
ASTERISK-09916: [patch] sometimes, cppflags or ldflags are missing
ASTERISK-09917: Segfault in do_monitor/ast_sched_runq
ASTERISK-09918: discrepacy in sendtext
ASTERISK-09919: discrepacy in sendtext
ASTERISK-09920: asterisk is just not a PBX
ASTERISK-09921: Large list of recording files produces delay on incoming calls on queues
ASTERISK-09922: [patch] Remove requirement of line@device on Dial() syntax with chan_skinny
ASTERISK-09923: [patch] Dialplan mutexes
ASTERISK-09924: socket_read() does not properly check for recvfrom() error
ASTERISK-09925: Use of reserved C++ terms
ASTERISK-09926: [patch] Allow the astdb in a userconfigurable directory
ASTERISK-09927: [patch] OpenBSD fix for chan_skinny.c
ASTERISK-09928: [patch] Replace killall with pkill
ASTERISK-09929: [patch] more res_agi updates, upgrades and fixes
ASTERISK-09930: More minor doxygen updates
ASTERISK-09931: Bridging channel hangs up during Bridge() application.
ASTERISK-09932: asterisk doesn't respect rtp port range
ASTERISK-09933: T.38 passthrough with 2 Asterisk boxes not working
ASTERISK-09934: Not possible to work with negative numbers as first argument to MATH
ASTERISK-09935: Core dump using 7921/7920 phones
ASTERISK-09936: [patch] replace ast_verbose(VERBOSE_PREFIX_X with ast_verb(X,
ASTERISK-09937: [patch] Make 'agi_network: yes' show up with agi debugging on (and handle partial writes)
ASTERISK-09938: ${ANSWEREDTIME}
ASTERISK-09939: does iax2 channels surpport group call pickup?
ASTERISK-09940: /var/log/asterisk/messages is filled with chan_zap.c: No D-channels available! Using Primary channel as D-channel anyway!
ASTERISK-09941: Misdn hang
ASTERISK-09942: LaTeX documentation update
ASTERISK-09943: D parameter in Dial() don't send DTMF to caller
ASTERISK-09944: Segmentation fault
ASTERISK-09945: [patch] [res_agi] more fixes and love, should be now completly thread and usecount safe.
ASTERISK-09946: Old LAGRQ frames showing up in new IAX2 calls
ASTERISK-09947: dtmf_end_only test failing on sip-iax bridge
ASTERISK-09948: Skinny to skinny calls one-way audio
ASTERISK-09949: Support reload for chan_skinny
ASTERISK-09950: Give more information if someone forgets to define loadzone and defaultzone
ASTERISK-09951: Calls drop with following debug error DEBUG[30271] channel.c: Didn't get a frame from channel: IAX2/724careo-16387
ASTERISK-09952: segfault on reload
ASTERISK-09953: [patch] Unload/load support for chan_skinny
ASTERISK-09954: [Patch] Post 1.4 Add custom Moh to app_queue on the fly
ASTERISK-09955: 7920 Phone Screen Not cleared after Call
ASTERISK-09956: memory corruption on freebsd sparc64
ASTERISK-09957: [patch] channel_find_locked causes loops (e.g. in handle_chanlist)
ASTERISK-09958: Wrong duration value in SIP INFO messages when sending DTMF
ASTERISK-09959: app_SpeechBackground system has variety of failures with DTMF #
ASTERISK-09960: Hint subscriptions do not update without peer call-limit entry
ASTERISK-09961: SIP peer re-register using Realtime causes to loop and segfault.
ASTERISK-09962: LaTeX documentation update
ASTERISK-09963: main/utils.c:ast_random() sometimes returns negative values which is illegal in SIP session ID
ASTERISK-09964: Invalid free of connection pointer
ASTERISK-09965: Chan SIP and ACL Source Based Routing
ASTERISK-09966: res_agi behaviour change in 1.2 & 1.4
ASTERISK-09967: Crash on ast_openstream on disconnected (at that moment) channel
ASTERISK-09968: Bug 10274 seems to be SIP related, not T.38. Please update.
ASTERISK-09969: chan_sip hangs with big number of sip channels
ASTERISK-09970: Hold Status Is Not Cleared On Blind Transfer (Re: #0010165)
ASTERISK-09971: 1.4.9 changes to app_queue breaks 'joinempty=yes'
ASTERISK-09972: [patch] Switch pbx_ael to generate code with comma delimiters
ASTERISK-09973: [PATCH] Device state shows Hold after SIP native transfer
ASTERISK-09974: chan_skinny randomly crashing asterisk
ASTERISK-09975: WaitExten doesn't playback MoH without class specified
ASTERISK-09976: Transfer Function not working
ASTERISK-09977: Asterisk crash for unknown reason
ASTERISK-09978: cant park a parked call
ASTERISK-09979: [patch] PCMA/16000 and PCMU/16000 support (hd telephony)
ASTERISK-09980: SIP - ACK not processed, 200 OK retransmitted
ASTERISK-09981: Asterisk is not sending the BYE
ASTERISK-09982: Hangup command is ignored when called within a macro that was called by using MacroIf.
ASTERISK-09983: SIP history is recorded for many (not all) items even when recording is off
ASTERISK-09984: directrtpsetup doesn't work with 'nat=yes' devices
ASTERISK-09985: [patch] change the output for "minivm show stats"
ASTERISK-09986: 1.2 version of : 0008943: inUse counter not decremented after hanging up a call which is on hold
ASTERISK-09987: On outgoing calls, failed reason does not get put into a variable
ASTERISK-09988: Blocking Asterisk is connection to server disappears
ASTERISK-09989: On Mac OS X PowerPC, Asterisk 1.4.x stops bridging new calls shortly after start
ASTERISK-09990: ZAP-channel terminates after 3 seconds with hangup cause 18
ASTERISK-09991: Asterisk crashed at res_musiconhold
ASTERISK-09992: Using dynamic realtime members of a queue can cause callers to wait forever
ASTERISK-09993: Asterisk Crashes in cdr_csv.c during csv_log while trying to close the log file
ASTERISK-09994: Certain realtime IAX calls are causing an malloc error and crash
ASTERISK-09995: [patch] Add the Ring time in the CONNECT on the queue_log and on the Manager event AgentConnect
ASTERISK-09996: i not have sound in ooh323
ASTERISK-09997: Help command in CLI dumps core under Solaris 10 X86
ASTERISK-09998: When I allow the codec G729 first in sip.conf the other codecs are not offered in the INVITE
ASTERISK-09999: [patch] Add Basic Support For RFC 4662 (Subscribe to lists)